gstreamer/gst-libs/gst/audio/audio.h
Thomas Vander Stichele 41585b7e78 moving and renaming we put the libs in the source in gst-libs/gst/(dir) the headers get installed in prefix/include/g...
Original commit message from CVS:
moving and renaming
we put the libs in the source in gst-libs/gst/(dir)
the headers get installed in prefix/include/gst/(dir)
the libs are installed in prefix/lib/gst
with a libgst prefix
the sources should be without the gst prefix
as per irc agreement
please comment if this sounds like a bad idea ;)
2001-12-22 23:43:34 +00:00

109 lines
4.6 KiB
C

/* Gnome-Streamer
* Copyright (C) <1999> Erik Walthinsen <omega@cse.ogi.edu>
* Library <2001> Thomas Vander Stichele <thomas@apestaart.org>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
* Boston, MA 02111-1307, USA.
*/
#include <gst/gst.h>
/* for people that are looking at this source: the purpose of these defines is
* to make GstCaps a bit easier, in that you don't have to know all of the
* properties that need to be defined. you can just use these macros. currently
* (8/01) the only plugins that use these are the passthrough, speed, volume,
* and [de]interleave plugins. so. these are for convenience only, and do not
* specify the 'limits' of gstreamer. you might also use these definitions as a
* base for making your own caps, if need be.
*
* for example, to make a source pad that can output mono streams of either
* float or int:
template = gst_padtemplate_new
("sink", GST_PAD_SINK, GST_PAD_ALWAYS,
gst_caps_append(gst_caps_new ("sink_int", "audio/raw",
GST_AUDIO_INT_PAD_TEMPLATE_PROPS),
gst_caps_new ("sink_float", "audio/raw",
GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS)),
NULL);
srcpad = gst_pad_new_from_template(template,"src");
* Andy Wingo, 18 August 2001 */
#define GST_AUDIO_INT_PAD_TEMPLATE_PROPS \
gst_props_new (\
"format", GST_PROPS_STRING ("int"),\
"law", GST_PROPS_INT (0),\
"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
"signed", GST_PROPS_LIST (\
GST_PROPS_BOOLEAN (TRUE),\
GST_PROPS_BOOLEAN(FALSE)\
),\
"width", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
"depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
"rate", GST_PROPS_INT_RANGE (4000, 96000),\
"channels", GST_PROPS_INT_RANGE (1, G_MAXINT),\
NULL)
#define GST_AUDIO_INT_MONO_PAD_TEMPLATE_PROPS \
gst_props_new (\
"format", GST_PROPS_STRING ("int"),\
"law", GST_PROPS_INT (0),\
"endianness", GST_PROPS_INT (G_BYTE_ORDER),\
"signed", GST_PROPS_LIST (\
GST_PROPS_BOOLEAN (TRUE),\
GST_PROPS_BOOLEAN(FALSE)\
),\
"width", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
"depth", GST_PROPS_LIST (GST_PROPS_INT(8), GST_PROPS_INT(16)),\
"rate", GST_PROPS_INT_RANGE (4000, 96000),\
"channels", GST_PROPS_INT (1),\
NULL)
#define GST_AUDIO_FLOAT_MONO_PAD_TEMPLATE_PROPS \
gst_props_new (\
"format", GST_PROPS_STRING ("float"),\
"layout", GST_PROPS_STRING ("gfloat"),\
"intercept", GST_PROPS_FLOAT (0.0),\
"slope", GST_PROPS_FLOAT (1.0),\
"rate", GST_PROPS_INT_RANGE (4000, 96000),\
"channels", GST_PROPS_INT (1),\
NULL)
/*
* this library defines and implements some helper functions for audio
* handling
*/
/* get byte size of audio frame (based on caps of pad */
int gst_audio_frame_byte_size (GstPad* pad);
/* get length in frames of buffer */
long gst_audio_frame_length (GstPad* pad, GstBuffer* buf);
/* get frame rate based on caps */
long gst_audio_frame_rate (GstPad *pad);
/* calculate length in seconds of audio buffer buf based on caps of pad */
double gst_audio_length (GstPad* pad, GstBuffer* buf);
/* calculate highest possible sample value based on capabilities of pad */
long gst_audio_highest_sample_value (GstPad* pad);
/* check if the buffer size is a whole multiple of the frame size */
gboolean gst_audio_is_buffer_framed (GstPad* pad, GstBuffer* buf);